5 files changed, 21 insertions, 31 deletions
diff --git a/src/encoder/flac_encoder.c b/src/encoder/flac_encoder.c
index e32588e2..d5b9ae26 100644
--- a/src/encoder/flac_encoder.c
+++ b/src/encoder/flac_encoder.c
@@ -173,21 +173,17 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 	/* FIXME: flac should support 32bit as well */
 	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S8:
+    case AV_SAMPLE_FMT_U8:
 		bits_per_sample = 8;
 		break;
 
-	case SAMPLE_FORMAT_S16:
+    case AV_SAMPLE_FMT_S16:
 		bits_per_sample = 16;
 		break;
 
-	case SAMPLE_FORMAT_S24_P32:
-		bits_per_sample = 24;
-		break;
-
 	default:
-		bits_per_sample = 24;
-		audio_format->format = SAMPLE_FORMAT_S24_P32;
+		bits_per_sample = 16;
+		audio_format->format = AV_SAMPLE_FMT_S16;
 	}
 
 	/* allocate the encoder */
@@ -291,24 +287,17 @@ flac_encoder_write(struct encoder *_encoder,
 	num_samples = num_frames * encoder->audio_format.channels;
 
 	switch (encoder->audio_format.format) {
-	case SAMPLE_FORMAT_S8:
+    case AV_SAMPLE_FMT_U8:
 		exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*4);
 		pcm8_to_flac(exbuffer, data, num_samples);
 		buffer = exbuffer;
 		break;
 
-	case SAMPLE_FORMAT_S16:
+    case AV_SAMPLE_FMT_S16:
 		exbuffer = pcm_buffer_get(&encoder->expand_buffer, length*2);
 		pcm16_to_flac(exbuffer, data, num_samples);
 		buffer = exbuffer;
 		break;
-
-	case SAMPLE_FORMAT_S24_P32:
-	case SAMPLE_FORMAT_S32:
-		/* nothing need to be done; format is the same for
-		   both mpd and libFLAC */
-		buffer = data;
-		break;
 	}
 
 	/* feed samples to encoder */
diff --git a/src/encoder/lame_encoder.c b/src/encoder/lame_encoder.c
index 3bb99ea2..e7c3b5a7 100644
--- a/src/encoder/lame_encoder.c
+++ b/src/encoder/lame_encoder.c
@@ -26,6 +26,8 @@
 #include <assert.h>
 #include <string.h>
 
+#include <libavutil/samplefmt.h>
+
 struct lame_encoder {
 	struct encoder encoder;
 
@@ -192,7 +194,7 @@ lame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 {
 	struct lame_encoder *encoder = (struct lame_encoder *)_encoder;
 
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format->format = AV_SAMPLE_FMT_S16;
 	audio_format->channels = 2;
 
 	encoder->audio_format = *audio_format;
diff --git a/src/encoder/twolame_encoder.c b/src/encoder/twolame_encoder.c
index 934b2ab2..cdbe1f1b 100644
--- a/src/encoder/twolame_encoder.c
+++ b/src/encoder/twolame_encoder.c
@@ -26,6 +26,8 @@
 #include <assert.h>
 #include <string.h>
 
+#include <libavutil/samplefmt.h>
+
 struct twolame_encoder {
 	struct encoder encoder;
 
@@ -192,7 +194,7 @@ twolame_encoder_open(struct encoder *_encoder, struct audio_format *audio_format
 {
 	struct twolame_encoder *encoder = (struct twolame_encoder *)_encoder;
 
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format->format = AV_SAMPLE_FMT_S16;
 	audio_format->channels = 2;
 
 	encoder->audio_format = *audio_format;
diff --git a/src/encoder/vorbis_encoder.c b/src/encoder/vorbis_encoder.c
index fcf2b513..728d1f76 100644
--- a/src/encoder/vorbis_encoder.c
+++ b/src/encoder/vorbis_encoder.c
@@ -28,6 +28,8 @@
 
 #include <assert.h>
 
+#include <libavutil/samplefmt.h>
+
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "vorbis_encoder"
 
@@ -213,7 +215,7 @@ vorbis_encoder_open(struct encoder *_encoder,
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
 	bool ret;
 
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format->format = AV_SAMPLE_FMT_S16;
 
 	encoder->audio_format = *audio_format;
 
diff --git a/src/encoder/wave_encoder.c b/src/encoder/wave_encoder.c
index 9eeb4d51..fe087659 100644
--- a/src/encoder/wave_encoder.c
+++ b/src/encoder/wave_encoder.c
@@ -26,6 +26,8 @@
 #include <assert.h>
 #include <string.h>
 
+#include <libavutil/samplefmt.h>
+
 struct wave_encoder {
 	struct encoder encoder;
 	unsigned bits;
@@ -114,24 +116,17 @@ wave_encoder_open(struct encoder *_encoder,
 	assert(audio_format_valid(audio_format));
 
 	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S8:
+    case AV_SAMPLE_FMT_U8:
 		encoder->bits = 8;
 		break;
-
-	case SAMPLE_FORMAT_S16:
+    case AV_SAMPLE_FMT_S16:
 		encoder->bits = 16;
 		break;
-
-	case SAMPLE_FORMAT_S24_P32:
-		encoder->bits = 24;
-		break;
-
-	case SAMPLE_FORMAT_S32:
+    case AV_SAMPLE_FMT_S32:
 		encoder->bits = 32;
 		break;
-
 	default:
-		audio_format->format = SAMPLE_FORMAT_S16;
+		audio_format->format = AV_SAMPLE_FMT_S16;
 		encoder->bits = 16;
 		break;
 	}